Відео

Hey Nathan, I'm about to do a theatre show on an X32. Have done a previous one with a Allen & Heath SQ5, which has great scene control, but didn't think the Behringer did the same thing. Your video was great and gave me all the tools and processes to be fully in control for this production. Thanks heaps!

you push the button, to keep the 5,9ms........you had another option to choose

can't you just apply the necessery crossover and than correct the acoustical response to the target with eq ? this way you can change slopes and cross filters freely

This was by far the most intuitive video about delays I could find on UA-cam. Thanks for sharing this publicly, a great resource! ❤

i was just wondering about this topic. thanks!

@Nathanlively, which program were you using for this video?

This was EXACTLY the explanation I was hoping I would find. I knew generally THAT this worked but hadnt yet see visually HOW. Thanks much.

estub buen

Sensacional!

any updates for my 69th birthday back here in 2024? thnx mnx

sorry if I have another question even if it's not the correct video. Do cardioid but in-line subwoofers give the same result? Of course if there is the possibility and you have enough space

Can we use 24dB L-R for the Sub. And 12dB BW for the Full, while the nature respond are not the same?

Is this being measured with a mic(s) only?

We are too thin-skinned these days. Go ahead and be a jerk if you can tell me the absolute best equipment, positioning, and set-up. Dude, we can take it.

you might want to use the notch method on a Monitor on stage. but clearly only because it's affect for the feedback loop while no effective change to listener. each monitor will be able to be independantly adjusted hard limiting the monitor would really end the deal guitars with mics can be used with ISO cabs. plexi shields for drummers etc etc before you really ever need to notch the PA channels... and if you did you might only need to notch the 120s at the bottom of the tower

hello and thanks for this video. I'm using a translator because I don't speak English well. I tried this configuration and noticed that when the public subwoofer is on the ground I get almost 2 dB more. my question is: is it right to always use this way or are there occasions when it is better to have the audience subwoofer on high?

can pretty much any top and sub be aligned using subaligner?

For those who did not know where the wrap around 10 milliseconds came from just like me. Frequency itself vibrates 100 cycles per second (100Hz). Therefore 1 Cycle should take 1/100 seconds to complete. While 1 second = 1000 milliseconds. 10ms is the time for frequency 100Hz to complete one cycle.

Yep, I just tried an active crossover with the Linkwitz filters, and it killed the sound coming out of my Maggies. Flat, lifeless, but the crossover worked. I went back to my non-Linkwitz pro audio crossover and we were back in business. I'm not a measurement guy, not with audio gear, for this very reason. So many folks go by measurements, just silly in my book, because in my experience, this is the result, crap sound.

Is Smaart software is free

What if the crossover is at 80Hz?

hi! we operate in a church campus, so we setup and strike down weekly. I didnt quite understand how to keep the Receivers stacked, without getting signal drops. You mentioned antenna about antenna splitters. how does that work? thank you

thank you

I didn't get a transition from an amplitude to a power... You imported an acoustical data and continued to work with electrical amplitude graphs. But.. When -3dB changed to -6dB pls?

What if we want a 3 line array instead of 2 ? Should we just reverse the 3rd one with no delay ? Making it kinda of a mix between cardioid and endfire ? Or you have a better idea ?

Hi ! How do you know if you're not aligning with a whole wavelengh delay (or 2 and so on...) ? Thanks.

What does the C mean in subaligner when you look at the result plot? …also Aligning Speakers that are phase compatible from the same manufacturer… could you recommend a video to watch on that

If you only have one mic, in my case a sonarworks reference ID RTA mic, would you suggest tuning my left right by just tuning one side (having mic as close to the speaker as possible and running pink through just that one speaker) and then copying and pasting the eq to both sides or would it be beneficial to run pink through both while being center to the two? Or would that create comb filtering?

Great video as usual Nathan, arguably the most well-explained all-pass filter usage in real alignment situation. Thank you!

Nathan, I just wanted to say thank you so very much for doing what you do and sharing all these wonderful and very informative videos, I know it's appreciated by many! I really like your style and please keep up the awesome work! Oh yeah, I too did the gas in a diesel truck thing once (hopefully never again) it ended costing me a little over a grand, still to this day I cannot pull up to a pump that has both diesel and gas without trembling from the traumatic experience and triple checking myself while filling up. Thanks again!

So we can give headphones to the listeners and make the gain 200db. What a wonderful world!

Thanks a lot for sharing this video! At first I was lazy to watch it because it lasted 1 hour, but when I started watching it I was hooked. It's exactly what I need at this moment. I have a project precisely to improve the GBF of the meeting space where I work, and all this information will help me to be able to explain it better to my colleagues and superiors.

Thank you.

Thanks for such a great video! I’m just starting to watch it. What about when the person on stage is whispering?

@nathanlively Thanks a lot for this material, got a few eye-opening things from it. Also, a big thanks to Mr Jason Romney, for sharing this unique approach, and for sharing their whole book for free. Please keep it up, it's a real value for a sound nerds like me, who's interested in how and why things work. You're awesome!

I use feedback regularly as my spectrum analyzer and it didn’t lie me ever. And also, since feedback is defined as a 1:1 ratio between the pressure arriving at the microphone and entering it, (imagine a electronic circuit in short, or a ring that closes its edges), it can be used to set the right volume on your amps to produce the right spl for the event.

Feedback doesn't occur when the sound from the loudspeaker reaches the microphone at the same level as it would directly from the person speaking. It happens when the waves from the loudspeaker can get back to the loudspeaker via the microphone at a time such that their peaks match the peaks already coming out of the loudspeaker. It's that constructive interference that causes the increase in volume of that frequency, and since it continues on the next round, that frequency builds to the familiar squeal. Being the same level doesn't matter; the waves being in sync does.

Literature says, that in reverberant rooms because of the many overlaying frequecies and thus combfiltering you have even less headroom for feedback occurance. Statistically beyind a specific frequency thats dependent on Roomvolume and T60 you statistically have the same Peak to Average Ratio in the Frequency Response which is 12 dB (Schroeder 1954, Schroeder & Kuttruff 1962, Ahnert & Reichert 1981). Kuttruff recommends even more Headroom on top for Speech (5 dB) and music (12dB) (Kuttruff & Hesselmann 1976). This results in a deduction in gain before feedback of 17dB for speech and 24dB for music in reverberant rooms. So the example in the video is only applicable for freefield.

Ok, but what are the odds that a person qualified to be employed as a sound engineer is going to show up to work & turn on a system in which the microphones are not going to be cardioid, the speakers are not already going to be positioned as optimum as the house will accommodate & the source is not already going to be at the optimum proximity to the microphone? Typically, when a person qualified for a position as a sound engineer resorts to using the EQ for stabilization, all the variables addressed in this video are as optimum as they are going to get &, essentially, any stabilization issues should only pertain to the stage monitors which, ideally, should be a secondary system isolated from the FOH system, but, even when they are a subsystem of FOH, the compromise in sonic quality for the sake of stabilization is tolerated. The fact of the matter is, in a professional real-world scenario, an engineer IS going to need to apply EQ'ing in an effort to achieve stability, but it should only pertain to the monitor system or subsystem... Ideally, aside from MAYBE a little roll off at the very top &/or very bottom of the spectrum & "flavoring to taste", if you will, the FOH EQ's, should be "flat"... Of course, the amount one may need to vary from "flat" to "flavor to taste" is relative to the inherent acoustic qualities of the venue... Or, if outdoors, the pertinent atmospheric conditions. That all been said... Three words: "In" "Ear" "Monitors"... "GBF" problem 𝗌̶𝗈̶𝗅̶𝗏̶𝖾̶𝖽̶...eliminated.

Another question is the issue of multiple mics. The more mics, the lower the available gain. Isn’t it something like every time you double the number of open mics, you lose 3db (or is it 6?) of gain before feedback? ?

On the question of why systems feedback when the band stops playing. I believe the most likely “culprit” is that all of your various compressors relax, effectively increasing the overall system gain. Remember, a compressor is a gain REDUCTION tool. When the signal goes below the threshold, it stops reducing gain. Most of us run compressors on individual channels and more on the various mix buses. All of those let loose when the signal level falls…overall gain goes up, feedback ensues. Gates and plugins like PSE can help combat this (along with mixing 😂)

Make a video for car audio measurements with single microphone 🎤 with smart and what setting keep while taking a measurement in car

I'm trying to learn better alignment in the field, especially as you say, sometimes with very limited time to run tests. Using 80hz as the example like you do in the video, what if I just reverse polarity of one of them, say the subs, run a test tone of 80 hz through both speakers. Then I add delay to the closest speaker to my listening location until I hear cancellation. Once I flip phase back, I should have summation, but am I also aligned (roughly)?

4 * 1.67 = 6.68